Android 音频开发（四）如何播放一帧音频数据下

再看这一篇文章前,如果你是小白，我建议你先看一下Android 音频开发（一）基础入门篇这一篇。今天继续讲解如何通过Android SDK自带API实现播放一帧音频数据。

我们都知道，Android SDK 自带了三种API实现播放一帧音频数据，他们分别是MediaPlayer，SoundPool，AudioTrack，如果你对这三种API不是很了解，可以先看我的上一篇Android 音频开发（三）如何播放一帧音频数据上这一篇，这里不在详细介绍。简单来说，MediaPlayer 更加适合在后台长时间播放本地音乐文件或者在线的流式资源; SoundPool 则适合播放比较短的音频片段，比如游戏声音、按键声、铃声片段等等，它可以同时播放多个音频; 而 AudioTrack 则更接近底层，提供了非常强大的控制能力，支持低延迟播放，适合流媒体和VoIP语音电话等场景。

音频的开发，更广泛地应用不仅仅局限于播放本地文件或者音频片段，因此，本文重点关注如何利AudioTrack API 来播放音频数据（注意，使用AudioTrack播放的音频必须是解码后的PCM数据）。

1. AudioTrack 的工作流程

在上一篇我们知道AudioTrack 的工作流程大致分如下四步：

配置参数，初始化内部的音频播放缓冲区
开始播放
需要一个线程，不断地向 AudioTrack 的缓冲区“写入”音频数据，注意，这个过程一定要及时，否则就会出现“underrun”的错误，该错误在音频开发中比较常见，意味着应用层没有及时地“送入”音频数据，导致内部的音频播放缓冲区为空。
停止播放，释放资源

2. AudioTrack 的参数配置

在看参数配配置前，我们先看看构造方法，然后一一介绍各个配置参数的意义：

    /*** Class constructor.* @param streamType the type of the audio stream. See*   {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM},*   {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC},*   {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}.* @param sampleRateInHz the initial source sample rate expressed in Hz.*   {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} means to use a route-dependent value*   which is usually the sample rate of the sink.*   {@link #getSampleRate()} can be used to retrieve the actual sample rate chosen.* @param channelConfig describes the configuration of the audio channels.*   See {@link AudioFormat#CHANNEL_OUT_MONO} and*   {@link AudioFormat#CHANNEL_OUT_STEREO}* @param audioFormat the format in which the audio data is represented.*   See {@link AudioFormat#ENCODING_PCM_16BIT},*   {@link AudioFormat#ENCODING_PCM_8BIT},*   and {@link AudioFormat#ENCODING_PCM_FLOAT}.* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is*   read from for playback. This should be a nonzero multiple of the frame size in bytes.*   <p> If the track's creation mode is {@link #MODE_STATIC},*   this is the maximum length sample, or audio clip, that can be played by this instance.*   <p> If the track's creation mode is {@link #MODE_STREAM},*   this should be the desired buffer size*   for the <code>AudioTrack</code> to satisfy the application's*   latency requirements.*   If <code>bufferSizeInBytes</code> is less than the*   minimum buffer size for the output sink, it is increased to the minimum*   buffer size.*   The method {@link #getBufferSizeInFrames()} returns the*   actual size in frames of the buffer created, which*   determines the minimum frequency to write*   to the streaming <code>AudioTrack</code> to avoid underrun.*   See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size*   for an AudioTrack instance in streaming mode.* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}* @throws java.lang.IllegalArgumentException* @deprecated use {@link Builder} or*   {@link #AudioTrack(AudioAttributes, AudioFormat, int, int, int)} to specify the*   {@link AudioAttributes} instead of the stream type which is only for volume control.*/public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,int bufferSizeInBytes, int mode)throws IllegalArgumentException {this(streamType, sampleRateInHz, channelConfig, audioFormat,bufferSizeInBytes, mode, AudioManager.AUDIO_SESSION_ID_GENERATE);}/*** Class constructor with audio session. Use this constructor when the AudioTrack must be* attached to a particular audio session. The primary use of the audio session ID is to* associate audio effects to a particular instance of AudioTrack: if an audio session ID* is provided when creating an AudioEffect, this effect will be applied only to audio tracks* and media players in the same session and not to the output mix.* When an AudioTrack is created without specifying a session, it will create its own session* which can be retrieved by calling the {@link #getAudioSessionId()} method.* If a non-zero session ID is provided, this AudioTrack will share effects attached to this* session* with all other media players or audio tracks in the same session, otherwise a new session* will be created for this track if none is supplied.* @param streamType the type of the audio stream. See*   {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM},*   {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC},*   {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}.* @param sampleRateInHz the initial source sample rate expressed in Hz.*   {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} means to use a route-dependent value*   which is usually the sample rate of the sink.* @param channelConfig describes the configuration of the audio channels.*   See {@link AudioFormat#CHANNEL_OUT_MONO} and*   {@link AudioFormat#CHANNEL_OUT_STEREO}* @param audioFormat the format in which the audio data is represented.*   See {@link AudioFormat#ENCODING_PCM_16BIT} and*   {@link AudioFormat#ENCODING_PCM_8BIT},*   and {@link AudioFormat#ENCODING_PCM_FLOAT}.* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is*   read from for playback. This should be a nonzero multiple of the frame size in bytes.*   <p> If the track's creation mode is {@link #MODE_STATIC},*   this is the maximum length sample, or audio clip, that can be played by this instance.*   <p> If the track's creation mode is {@link #MODE_STREAM},*   this should be the desired buffer size*   for the <code>AudioTrack</code> to satisfy the application's*   latency requirements.*   If <code>bufferSizeInBytes</code> is less than the*   minimum buffer size for the output sink, it is increased to the minimum*   buffer size.*   The method {@link #getBufferSizeInFrames()} returns the*   actual size in frames of the buffer created, which*   determines the minimum frequency to write*   to the streaming <code>AudioTrack</code> to avoid underrun.*   You can write data into this buffer in smaller chunks than this size.*   See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size*   for an AudioTrack instance in streaming mode.* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}* @param sessionId Id of audio session the AudioTrack must be attached to* @throws java.lang.IllegalArgumentException* @deprecated use {@link Builder} or*   {@link #AudioTrack(AudioAttributes, AudioFormat, int, int, int)} to specify the*   {@link AudioAttributes} instead of the stream type which is only for volume control.*/public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,int bufferSizeInBytes, int mode, int sessionId)throws IllegalArgumentException {// mState already == STATE_UNINITIALIZEDthis((new AudioAttributes.Builder()).setLegacyStreamType(streamType).build(),(new AudioFormat.Builder()).setChannelMask(channelConfig).setEncoding(audioFormat).setSampleRate(sampleRateInHz).build(),bufferSizeInBytes,mode, sessionId);deprecateStreamTypeForPlayback(streamType, "AudioTrack", "AudioTrack()");}/*** Class constructor with {@link AudioAttributes} and {@link AudioFormat}.* @param attributes a non-null {@link AudioAttributes} instance.* @param format a non-null {@link AudioFormat} instance describing the format of the data*     that will be played through this AudioTrack. See {@link AudioFormat.Builder} for*     configuring the audio format parameters such as encoding, channel mask and sample rate.* @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is*   read from for playback. This should be a nonzero multiple of the frame size in bytes.*   <p> If the track's creation mode is {@link #MODE_STATIC},*   this is the maximum length sample, or audio clip, that can be played by this instance.*   <p> If the track's creation mode is {@link #MODE_STREAM},*   this should be the desired buffer size*   for the <code>AudioTrack</code> to satisfy the application's*   latency requirements.*   If <code>bufferSizeInBytes</code> is less than the*   minimum buffer size for the output sink, it is increased to the minimum*   buffer size.*   The method {@link #getBufferSizeInFrames()} returns the*   actual size in frames of the buffer created, which*   determines the minimum frequency to write*   to the streaming <code>AudioTrack</code> to avoid underrun.*   See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size*   for an AudioTrack instance in streaming mode.* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}.* @param sessionId ID of audio session the AudioTrack must be attached to, or*   {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction*   time. See also {@link AudioManager#generateAudioSessionId()} to obtain a session ID before*   construction.* @throws IllegalArgumentException*/public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,int mode, int sessionId)throws IllegalArgumentException {this(attributes, format, bufferSizeInBytes, mode, sessionId, false /*offload*/,ENCAPSULATION_MODE_NONE, null /* tunerConfiguration */);}private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,int mode, int sessionId, boolean offload, int encapsulationMode,@Nullable TunerConfiguration tunerConfiguration)throws IllegalArgumentException {super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);// mState already == STATE_UNINITIALIZEDmConfiguredAudioAttributes = attributes; // object copy not needed, immutable.if (format == null) {throw new IllegalArgumentException("Illegal null AudioFormat");}// Check if we should enable deep buffer modeif (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {mAttributes = new AudioAttributes.Builder(mAttributes).replaceFlags((mAttributes.getAllFlags()| AudioAttributes.FLAG_DEEP_BUFFER)& ~AudioAttributes.FLAG_LOW_LATENCY).build();}// remember which looper is associated with the AudioTrack instantiationLooper looper;if ((looper = Looper.myLooper()) == null) {looper = Looper.getMainLooper();}int rate = format.getSampleRate();if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {rate = 0;}int channelIndexMask = 0;if ((format.getPropertySetMask()& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {channelIndexMask = format.getChannelIndexMask();}int channelMask = 0;if ((format.getPropertySetMask()& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {channelMask = format.getChannelMask();} else if (channelIndexMask == 0) { // if no masks at all, use stereochannelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT| AudioFormat.CHANNEL_OUT_FRONT_RIGHT;}int encoding = AudioFormat.ENCODING_DEFAULT;if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {encoding = format.getEncoding();}audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);mOffloaded = offload;mStreamType = AudioSystem.STREAM_DEFAULT;audioBuffSizeCheck(bufferSizeInBytes);mInitializationLooper = looper;if (sessionId < 0) {throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);}int[] sampleRate = new int[] {mSampleRate};int[] session = new int[1];session[0] = sessionId;// native initializationint initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,offload, encapsulationMode, tunerConfiguration);if (initResult != SUCCESS) {loge("Error code "+initResult+" when initializing AudioTrack.");return; // with mState == STATE_UNINITIALIZED}mSampleRate = sampleRate[0];mSessionId = session[0];// TODO: consider caching encapsulationMode and tunerConfiguration in the Java object.if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) != 0) {int frameSizeInBytes;if (AudioFormat.isEncodingLinearFrames(mAudioFormat)) {frameSizeInBytes = mChannelCount * AudioFormat.getBytesPerSample(mAudioFormat);} else {frameSizeInBytes = 1;}mOffset = ((int) Math.ceil(HEADER_V2_SIZE_BYTES / frameSizeInBytes)) * frameSizeInBytes;}if (mDataLoadMode == MODE_STATIC) {mState = STATE_NO_STATIC_DATA;} else {mState = STATE_INITIALIZED;}baseRegisterPlayer();}/*** A constructor which explicitly connects a Native (C++) AudioTrack. For use by* the AudioTrackRoutingProxy subclass.* @param nativeTrackInJavaObj a C/C++ pointer to a native AudioTrack* (associated with an OpenSL ES player).* IMPORTANT: For "N", this method is ONLY called to setup a Java routing proxy,* i.e. IAndroidConfiguration::AcquireJavaProxy(). If we call with a 0 in nativeTrackInJavaObj* it means that the OpenSL player interface hasn't been realized, so there is no native* Audiotrack to connect to. In this case wait to call deferred_connect() until the* OpenSLES interface is realized.*//*package*/ AudioTrack(long nativeTrackInJavaObj) {super(new AudioAttributes.Builder().build(),AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);// "final"smNativeTrackInJavaObj = 0;mJniData = 0;// remember which looper is associated with the AudioTrack instantiationLooper looper;if ((looper = Looper.myLooper()) == null) {looper = Looper.getMainLooper();}mInitializationLooper = looper;// other initialization...if (nativeTrackInJavaObj != 0) {baseRegisterPlayer();deferred_connect(nativeTrackInJavaObj);} else {mState = STATE_UNINITIALIZED;}}

1:streamType

这个参数代表着当前应用使用的哪一种音频管理策略，当系统有多个进程需要播放音频时，这个管理策略会决定最终的展现效果，该参数的可选的值以常量的形式定义在 AudioManager 类中，主要包括：

STREAM_VOCIE_CALL：电话声音

STREAM_SYSTEM：系统声音

STREAM_RING：铃声

STREAM_MUSCI：音乐声

STREAM_ALARM：警告声

STREAM_NOTIFICATION：通知声采样率，从AudioTrack源码的“audioParamCheck”函数可以看到，这个采样率的取值范围必须在 4000Hz～192000Hz 之间。

2:sampleRateInHz

采样率，从AudioTrack源码的“audioParamCheck”函数可以看到，这个采样率的取值范围必须在 4000Hz～192000Hz 之间。

3:channelConfig

通道数的配置，可选的值以常量的形式定义在 AudioFormat 类中，常用的是 CHANNEL_IN_MONO（单通道），CHANNEL_IN_STEREO（双通道）

4:audioFormat

这个参数是用来配置“数据位宽”的，可选的值也是以常量的形式定义在 AudioFormat 类中，常用的是 ENCODING_PCM_16BIT（16bit），ENCODING_PCM_8BIT（8bit），注意，前者是可以保证兼容所有Android手机的。

5:bufferSizeInBytes

这个是最难理解又最重要的一个参数，它配置的是 AudioTrack 内部的音频缓冲区的大小，该缓冲区的值不能低于一帧“音频帧”（Frame）的大小，而前一篇文章介绍过，一帧音频帧的大小计算如下：

int size = 采样率 x 位宽 x 采样时间 x 通道数

采样时间一般取 2.5ms~120ms 之间，由厂商或者具体的应用决定，我们其实可以推断，每一帧的采样时间取得越短，产生的延时就应该会越小，当然，碎片化的数据也就会越多。

在Android开发中，AudioTrack 类提供了一个帮助你确定这个 bufferSizeInBytes 的函数，原型如下：

int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat);

不同的厂商的底层实现是不一样的，但无外乎就是根据上面的计算公式得到一帧的大小，音频缓冲区的大小则必须是一帧大小的2～N倍，有兴趣的朋友可以继续深入源码探究探究。

实际开发中，强烈建议由该函数计算出需要传入的 bufferSizeInBytes，而不是自己手动计算。

6:mode

AudioTrack 提供了两种播放模式，一种是 static 方式，一种是 streaming 方式，前者需要一次性将所有的数据都写入播放缓冲区，简单高效，通常用于播放铃声、系统提醒的音频片段; 后者则是按照一定的时间间隔不间断地写入音频数据，理论上它可用于任何音频播放的场景。

可选的值以常量的形式定义在 AudioTrack 类中，一个是 MODE_STATIC，另一个是 MODE_STREAM，根据具体的应用传入对应的值即可。

3. AudioTrack 的封装

下面是我对AudioTrack 类的接口简单封装了一下，提供了一个 AudioPlayer 工具类，先贴出代码如下：

package com.bnd.myaudioandvideo.utilsimport android.media.AudioFormat
import android.media.AudioManager
import android.media.AudioRecord
import android.media.AudioTrack
import android.os.Process
import java.io.DataInputStream
import java.io.File
import java.io.FileInputStreamclass AudioTrackManager {private var mAudioTrack: AudioTrack? = nullprivate var mDis //播放文件的数据流: DataInputStream? = nullprivate var mRecordThread: Thread? = nullprivate var isStart = false//指定缓冲区大小。调用AudioRecord类的getMinBufferSize方法可以获得。private var mMinBufferSize = 0private fun initData() {//根据采样率，采样精度，单双声道来得到frame的大小。mMinBufferSize = AudioTrack.getMinBufferSize(mSampleRateInHz, mChannelConfig, mAudioFormat) //计算最小缓冲区//注意，按照数字音频的知识，这个算出来的是一秒钟buffer的大小。//创建AudioTrackmAudioTrack = AudioTrack(mStreamType, mSampleRateInHz, mChannelConfig,mAudioFormat, mMinBufferSize, mMode)}/*** 销毁线程方法*/private fun destroyThread() {try {isStart = falseif (null != mRecordThread && Thread.State.RUNNABLE == mRecordThread!!.state) {try {Thread.sleep(500)mRecordThread!!.interrupt()} catch (e: Exception) {mRecordThread = null}}mRecordThread = null} catch (e: Exception) {e.printStackTrace()} finally {mRecordThread = null}}/*** 启动播放线程*/private fun startThread() {destroyThread()isStart = trueif (mRecordThread == null) {mRecordThread = Thread(recordRunnable)mRecordThread!!.start()}}/*** 播放线程*/var recordRunnable = Runnable {try {//设置线程的优先级Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO)val tempBuffer = ByteArray(mMinBufferSize)var readCount = 0while (mDis!!.available() > 0) {readCount = mDis!!.read(tempBuffer)if (readCount == AudioTrack.ERROR_INVALID_OPERATION || readCount == AudioTrack.ERROR_BAD_VALUE) {continue}if (readCount != 0 && readCount != -1) { //一边播放一边写入语音数据//判断AudioTrack未初始化，停止播放的时候释放了，状态就为STATE_UNINITIALIZEDif (mAudioTrack!!.state == AudioTrack.STATE_UNINITIALIZED) {initData()}mAudioTrack!!.play()mAudioTrack!!.write(tempBuffer, 0, readCount)}}stopPlay() //播放完就停止播放} catch (e: Exception) {e.printStackTrace()}}/*** 播放文件* @param path* @throws Exception*/@Throws(Exception::class)private fun setPath(path: String) {val file = File(path)mDis = DataInputStream(FileInputStream(file))}/*** 启动播放** @param path*/fun startPlay(path: String) {try {
//            //AudioTrack未初始化
//            if(mAudioTrack.getState() == AudioTrack.STATE_UNINITIALIZED){
//                throw new RuntimeException("The AudioTrack is not uninitialized");
//            }//AudioRecord.getMinBufferSize的参数是否支持当前的硬件设备
//            else if (AudioTrack.ERROR_BAD_VALUE == mMinBufferSize || AudioTrack.ERROR == mMinBufferSize) {
//                throw new RuntimeException("AudioTrack Unable to getMinBufferSize");
//            }else{setPath(path)startThread()//            }} catch (e: Exception) {e.printStackTrace()}}/*** 停止播放*/fun stopPlay() {try {destroyThread() //销毁线程if (mAudioTrack != null) {if (mAudioTrack!!.state == AudioRecord.STATE_INITIALIZED) { //初始化成功mAudioTrack!!.stop() //停止播放}if (mAudioTrack != null) {mAudioTrack!!.release() //释放audioTrack资源}}if (mDis != null) {mDis!!.close() //关闭数据输入流}} catch (e: Exception) {e.printStackTrace()}}companion object {@Volatileprivate var mInstance: AudioTrackManager? = null//音频流类型private const val mStreamType = AudioManager.STREAM_MUSIC//指定采样率 （MediaRecoder 的采样率通常是8000Hz AAC的通常是44100Hz。 设置采样率为44100，目前为常用的采样率，官方文档表示这个值可以兼容所有的设置）private const val mSampleRateInHz = 44100//指定捕获音频的声道数目。在AudioFormat类中指定用于此的常量private const val mChannelConfig = AudioFormat.CHANNEL_CONFIGURATION_MONO //单声道//指定音频量化位数 ,在AudioFormaat类中指定了以下各种可能的常量。通常我们选择ENCODING_PCM_16BIT和ENCODING_PCM_8BIT PCM代表的是脉冲编码调制，它实际上是原始音频样本。//因此可以设置每个样本的分辨率为16位或者8位，16位将占用更多的空间和处理能力,表示的音频也更加接近真实。private const val mAudioFormat = AudioFormat.ENCODING_PCM_16BIT//STREAM的意思是由用户在应用程序通过write方式把数据一次一次得写到audiotrack中。这个和我们在socket中发送数据一样，// 应用层从某个地方获取数据，例如通过编解码得到PCM数据，然后write到audiotrack。private const val mMode = AudioTrack.MODE_STREAM/*** 获取单例引用** @return*/val instance: AudioTrackManager?get() {if (mInstance == null) {synchronized(AudioTrackManager::class.java) {if (mInstance == null) {mInstance = AudioTrackManager()}}}return mInstance}}init {initData()}
}

好了,至此AudioTrack的简单封装已经完毕，代码中也有详细的注释，如有不懂的地方，或者有何指教的地方欢迎大家在下方留言。

4. 总结

音频开发的知识点还是很多的，学习音频开发需要大家有足够的耐心，一步一个脚印的积累，只有这样才能把音频开发学好。下面推荐几个比较好的博主，希望对大家有所帮助。

csdn博主：《雷神雷霄骅》
51CTO博客：《Jhuster的专栏》